Topic: Call Recording - where are the files stored?

[fred2k3]

If I enable Call Recording System Default where does it store the files?

I have poked around /var/lib/asterisk and /home/e-smith and can't see anything.

[SARK devs]

/var/spool/asterisk/monitor, if you are using vanilla Asterisk

/var/spool/asterisk/monout or /var/spool/asterisk/monstage depending upon release and setup if you are using sail
Kind Regards
S

[fred2k3]

Great found them - thanks.

It doesn't seem to record incoming calls through our SIP trunks though. I've tried the setting on 'Incoming', 'Both' and 'OTR' (what does OTR stand for by the way?). On 'Both' it is recording internal calls, and outgoing calls but not incoming from external sources.

[SARK devs]

what release do you have?

[fred2k3]

3.1.1-22

[fred2k3]

Does anyone else have problems recording incoming calls?

[SARK devs]

Hi there

There aren't any known problems with call recording in 3.1.1-22.   If you send me a console log of an inbound call arriving then I may be able to help you further.   

Kind Regards
S

[fred2k3]

Here's an inbound call:

Code: [Select]

[Mar 17 17:43:17] WARNING[31257]: chan_sip.c:6343 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)
[Mar 17 17:43:17] WARNING[31257]: chan_sip.c:6343 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)
    -- Connected line update to DAHDI/i1/07754278298-1d prevented.
    -- SIP/6005-00000156 answered DAHDI/i1/07754278298-1d
    -- Executing [h@extensions:1] Hangup("DAHDI/i1/077########-1d", "") in new stack
  == Spawn extension (extensions, h, 1) exited non-zero on 'DAHDI/i1/07754278298-1d'
    -- <DAHDI/i1/07754278298-1d>AGI Script sarkhpe completed, returning 0
    -- Auto fallthrough, channel 'DAHDI/i1/07754278298-1d' status is 'ANSWER'
    -- Hungup 'DAHDI/i1/077########-1d'

No idea if this has any relevance, but there were hundreds of those "WARNING[31257]: chan_sip.c:6343 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write...." warnings messages and they clogged up the CLI so quickly I couldn't get the start of the trace. They were only present whilst our phones were ringing and stopped once the call was answered. These warnings are also there even if I turn call recording to None, so I don't think it's related to the call recording issue.

[SARK devs]

The warnings are caused by a codec mismatch between the phone (or MOH) and the ISDN trunk.   Likely you have the phones set to force G729 so the PBX has to translate.

I need to see the beginning of the trace to see if recording is started.  IN the meantime, check that the phones, and any included call groups, are all set to a recording  preference of default (assuming you have set BOTH in globals).

Cheers
S

[fred2k3]

Sorry for tardy response...

Extensions and Call groups both have recording options set to default, and yes Globals is set to BOTH.

I'm trying to eradicate the warning messages so I can get you the start of the trace. These are our normal asterisk codec config settings:

disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw

I tried commenting out ;allow=g729 on my extension, then rang it, but I still got all the WARNING messages when I answered my extension. How can I stop forcing G729? Or is there any way to stop printing the WARNING messages?

[SARK devs]

Hi,
I think it may be the phone which is attempting to impose G729 as its first choice.

Best

S

[fred2k3]

It's been a while.. but anyway I've got the start of the log file for an incoming call.

To recap, I'm trying to record calls - it works fine for outgoing and stores them in /var/spool/asterisk/monout as wav files, but no sign of the incoming calls. I have the Call Recording System Default set to BOTH and all extensions and call groups have the recording option set to default.

Code: [Select]

== Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.8.7.0 currently running on voip01 (pid = 4068)
voip01*CLI>
Verbosity is at least 28

voip01*CLI>
    -- Accepting call from '07760278491' to '741389' on channel 0/1, span 1
    -- Executing [741389@from-pstn:1] AGI("DAHDI/i1/07760278491-363", "sarkhpe,Inbound,741389,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe

voip01*CLI>
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=741389)

voip01*CLI>
    -- AGI Script Executing Application: (Wait) Options: (1)

voip01*CLI>
    -- Playing 'usergreeting0001' (escape_digits=123) (sample_offset 0)

voip01*CLI>
    -- <DAHDI/i1/07760278491-363>AGI Script sarkhpe completed, returning 0
    -- Executing [1002@extensions:1] AGI("DAHDI/i1/07760278491-363", "sarkhpe,Alias,SIP/6001 SIP/6002 SIP/6003 SIP/6005 SIP/6006 SIP/6007 SIP/6051,1002,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe

voip01*CLI>
    -- AGI Script Executing Application: (Set) Options: (CALLERID(rdnis)=1002)

voip01*CLI>
    -- AGI Script Executing Application: (Set) Options: (__filename=1409829441-1002-07760278491.wav)

voip01*CLI>
    -- AGI Script Executing Application: (Set) Options: (__channame=DAHDI/i1/07760278491-363)

voip01*CLI>
    -- AGI Script Executing Application: (Monitor) Options: (wav,1409829441-1002-07760278491,mb)

voip01*CLI>
    -- AGI Script Executing Application: (Dial) Options: (SIP/6001&SIP/6002&SIP/6003&SIP/6005&SIP/6006&SIP/6007&SIP/6051,40,ctI)
  == Using SIP RTP CoS mark 5
    -- Called SIP/6001
  == Using SIP RTP CoS mark 5
    -- Called SIP/6002
  == Using SIP RTP CoS mark 5
    -- Called SIP/6003
  == Using SIP RTP CoS mark 5
    -- Called SIP/6005
  == Using SIP RTP CoS mark 5
    -- Called SIP/6006
  == Using SIP RTP CoS mark 5
    -- Called SIP/6007
  == Using SIP RTP CoS mark 5
    -- Called SIP/6051
    -- Connected line update to DAHDI/i1/07760278491-363 prevented.
    -- Connected line update to DAHDI/i1/07760278491-363 prevented.
    -- Connected line update to DAHDI/i1/07760278491-363 prevented.
    -- Connected line update to DAHDI/i1/07760278491-363 prevented.
    -- Connected line update to DAHDI/i1/07760278491-363 prevented.
    -- Connected line update to DAHDI/i1/07760278491-363 prevented.
    -- Connected line update to DAHDI/i1/07760278491-363 prevented.
voip01*CLI>


[Sep  4 12:17:21] WARNING[29883]: chan_sip.c:6343 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)

[SARK devs]

What do you see in /var/spool/asterisk/monstage?

Kind Regards

S

[fred2k3]

...I see all my incoming calls Sorry, now feeling suitably sheepish Thanks for your help.

One more thing, can you explain the file naming convention please?

1409909483-1002-07956156127.wav

1409 at the start is obviously the year & month, then after the hyphen is the call group, and after the last hyphen is the caller's number, but what is "909483" ?  Doesn't seem to relate to the day or the time like I'd expect.

[SARK devs]

HI

No need to feel sheepish, call recording in Asterisk (and hence SARK) is complex.   We use different recording techniques for inbound and outbound because we need to be able to do slightly different things.  Upshot is that in the absence of the advanced call recording feature, which we sell to our paying clients, your outbound call recordings will end up in monout and your inbound call recordings will be in monstage.

The first string in the name is the time stamp in linux format, sometimes called the epoch.   You can fetch it to human readable format with the date command

Code: [Select]

date -d @1409909483
Fri Sep  5 10:31:23 BST 2014
My TZ is UK so it displays as British Daylight Saver Time (BST), yours will be different if you are in a different TZ

Kind Regards

S